Analysis of the latency of SIP phone based on the embedded system

Recently, the rapid development of VoIP has attracted considerable attention. However, VoIP service exposes weakness on voice quality such as delay and jitter. So it is helpful to improve the quality of VoIP service by mitigating time latency and jitter. In this paper, VoIP phone is implemented by open source software components using the Session Initiation Protocol (SIP) based on embedded systems. Then we analyze the reasons causing delay during the conversation between two VoIP phones which are implemented using open source software components, and propose a new scheme in which we increase the frequency of DMA buffer triggering interruption to reduce the latency, in accordance with the characteristics of the hardware. Finally, the experimental results verify the feasibility of the work.